Hi this is the macro used for that purpose ..
[macro-dialout-trunk] exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) ;check for CID override for exten exten => s,2,SetCallerID(${ECID${CALLERIDNUM}}) exten => s,3,Goto(6) exten => s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) ;check for CID override for trunk exten => s,5,SetCallerID(${OUTCID_${ARG1}}) exten => s,6,SetGroup(OUT_${ARG1}) exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}}) ; if we've used up the max channels, continue at 108 (n+101) exten => s,8,SetVar(DIAL_NUMBER=${ARG2}) exten => s,9,SetVar(DIAL_TRUNK=${ARG1}) exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial string for this trunk exten => s,11,Dial(${OUT_${ARG1}}/${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; if dial fails (ie, all channels are busy), continue at 112 (n+101) ;exten => s,11,Dial(Zap/0/${DIAL_NUMBER}) ; we should only get here if the call was successful (?) exten => s,9,Congestion ; exit points for macro exten => s,108,NoOp(max channels used up) exten => s,112,NoOp(dial failed) as u can see is also a dial instruction the call seems to be done but in fact my analog extension does not ring :/ any clue? Thanks again El mar, 03-05-2005 a las 10:17 -0500, Moises Silva escribió: > Hi Julio. It would be nice if you show the extensions.conf that > handles that kind of calls. You can do something like this: > > [macro-analogpbx] > exten => s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes > from other Zap ch > exten => s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3, > othewise 6 > exten => s,3,Flash() > exten => s,4,SendDTMF(${analogprefix}${num}) //send the DTMF for the > extension dialed > exten => s,5,Hangup() > exten => s,6,Dial(Zap/g${analoggroup}/${analogprefix}${num}) //if the > call comes from SIP or IAX then execute Dial trough some group in > zapata > exten => s,7,Hangup() > > You can see some variables i just use for administration of my PBX, > but i hope you understand the concept. > > Good Look > > - moy > > On 5/3/05, Julio Saura <[EMAIL PROTECTED]> wrote: > > Hi there > > > > i have an asterisk box running ok, and now i am trying to integrate it > > with my local analog pbx > > > > So far, i have connected the fxo port of my * to an analog extension > > port of my analog pbx. > > > > As far as i know, if a call an extension of my analog pbx on a sip phone > > ( i have done the right dial plan for routing these calls to de zap > > channel ) the analog pbx extension should ring ... > > > > am i right? > > > > asterisk says the call is done, but the analog extension keeps in > > silence .. :? > > > > any clue, am i doing something wrong? > > > > Best regards. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users